Log - 8eca1ff5107e274b7ac16cbb132f6b6f7c754597 - platform/external/webrtc

970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
ed7b4ff Use isolated-script-test-perf-output on low_bandwidth_audio_test. by Edward Lemur · 7 years ago
06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
d8b041c Ignore extra arguments in low_bandwidth_audio_test. by Edward Lemur · 7 years ago
649a385 Removes usage of analog AGC. by henrika · 7 years ago
90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
98d4036 Make it possible to run low_bandwidth_audio_test on Android swarming. by Edward Lemur · 7 years ago
b401771 Store JSON perf results for low_bandwidth_audio_test. by Edward Lemur · 7 years ago
8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
731082c Reland: Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
5a25ab2 Revert "Add mock_rtc_event_log.h." by Edward Lemur · 7 years ago
63aea46 Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
94dc177 Add mock_bitrate_controller.h. by Patrik Höglund · 7 years ago
6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
5e849cf Stop using public_deps in audio/utility. by Mirko Bonadei · 7 years ago
56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
63e6072 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. by Fredrik Solenberg · 7 years ago
2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
6d85252 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection AP (follow-up) by henrika · 7 years ago
675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
e4be4b7 Revert "Remove const from ThreadChecker in NullAudioPoller." by Mirko Bonadei · 7 years ago
54e41dd Remove const from ThreadChecker in NullAudioPoller. by Bjorn Terelius · 7 years ago
90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
9155e49 New classes RefCounter and RefCountedBase. by Niels Möller · 7 years ago
78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
6f72f56 Change return types of refcount methods. by Niels Möller · 7 years ago
dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
88b23f6 Fix flag name in low_bandwidth_audio_test.py by Edward Lemur · 7 years ago
7e3b569 Ignore swarming arguments in low_bandwidth_audio_test.py by Edward Lemur · 7 years ago
b0250f0 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
90e1f53 Fix potentional race in AudioSendStream constructor by Danil Chapovalov · 7 years ago
c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
45a0b36 Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
f4898a6 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
bb1222f Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
2019698 Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script. by Edward Lemur · 7 years ago
2011075 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. by Edward Lemur · 7 years ago
b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
4652e86 Disable flaky AudioStats.NoLoss test. by solenberg · 7 years ago
9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
5a6aa4f Fix path to root in low_bandwidth_audio_test.py by Henrik Kjellander · 7 years ago
92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago